[asterisk-users] SIP over TCP/TLS for 1.4 branch
Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Bryan Ekelund WHI Solutions, Inc. bekel...@whisolutions.com STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch
And take the easy way out? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, November 23, 2009 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Just file a bug with Microsoft and ask them to support SIP over UDP. Problem solved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Crash - ODBC Realtime
Running 1.4.26.2 on CentOS 5.3 Using ODBC with mysql for voicemail storage. Everytime my dialplan tries to open a connection to save a message or retrieve a stored message, Asterisk dumps out and restarts with: Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed It restarts fine, but still no voicemail. Nothing in a Mysql query log show anything of any interest either. Switching back to 'flat file' storage works fine. Any thoughts? Thanks. Bryan Ekelund STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Crash - ODBC Realtime
Upon further review, it is not dumping out, just restarting on its own with the same error. No .dmp in /tmp -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, September 21, 2009 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Crash - ODBC Realtime On Monday 21 September 2009 01:46:55 pm Ekelund, Bryan wrote: Running 1.4.26.2 on CentOS 5.3 Using ODBC with mysql for voicemail storage. Everytime my dialplan tries to open a connection to save a message or retrieve a stored message, Asterisk dumps out and restarts with: Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed It restarts fine, but still no voicemail. Nothing in a Mysql query log show anything of any interest either. Switching back to 'flat file' storage works fine. Any thoughts? Please read doc/backtrace.txt. You've given just enough information to verify that there is a problem, but not enough to diagnose where the problem lies. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Crash - ODBC Realtime
Running with -f -vvvg -c After recompiling with DONT_OPTIMIZE I no longer get: Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed but it sends an automatic restart, no core dump. As soon as it hits VoiceMail(x...@extensions,u) I get the disconnect message. Set the debug to 10. The last messages I get is : DEBUG[4522]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM voicemail_users WHERE mailbox = '9550' AND context = 'extensions' Disconnected from Asterisk server -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Monday, September 21, 2009 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Crash - ODBC Realtime Ekelund, Bryan escribió: Upon further review, it is not dumping out, just restarting on its own with the same error. No .dmp in /tmp Check that you are running asterisk with the -g option. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting multiple office with multiple servers
Greetings all, I currently manage a two-server asterisk system that connects two of our offices. Running 1.4 on CentOS 5.2 on both sides. We use Polycom 501 phone and register the phones to both systems, and use SIP peering to interconnect the two systems. We had been using IAX, but found that for some reason we were having trouble keeping that data stream in out QOS. Sometime in the near future, we are planning on integrating two of our other remote offices, each with their own asterisk server, into this network. I would like to have the phones register to two servers, but be able to be seen by all four. I have been experimenting with OpenSips/Kamailio as a registration server and forwarding all SIP requests to the appropriate office, but that may have a larger learning curve than I would like for the timeframe I am working with. I am looking at DUNDi and am thinking that this might be the way to merge these systems together and share the registrations between the servers. I am sure someone has experience with this type of setup, and I was hoping that I could confirm that DUNDi might be the way to go, or if not, maybe point me in the right direction. Thanks! Bryan Ekelund STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting multiple office with multiple servers
That was my intent with integrating Kamailio into the project. While I am fairly familiar with asterisk at this point, I am flailing around with Kamailio and taking a look at other options. I don't know if I will be comfortable putting that into a production environment with the amount of experience I have with it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Tuesday, July 21, 2009 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connecting multiple office with multiple servers On Tue, 2009-07-21 at 09:46 -0400, Ekelund, Bryan wrote: Greetings all, I currently manage a two-server asterisk system that connects two of our offices. Running 1.4 on CentOS 5.2 on both sides. We use Polycom 501 phone and register the phones to both systems, and use SIP peering to interconnect the two systems. We had been using IAX, but found that for some reason we were having trouble keeping that data stream in out QOS. Sometime in the near future, we are planning on integrating two of our other remote offices, each with their own asterisk server, into this network. I would like to have the phones register to two servers, but be able to be seen by all four. I have been experimenting with OpenSips/Kamailio as a registration server and forwarding all SIP requests to the appropriate office, but that may have a larger learning curve than I would like for the timeframe I am working with. I am looking at DUNDi and am thinking that this might be the way to merge these systems together and share the registrations between the servers. I am sure someone has experience with this type of setup, and I was hoping that I could confirm that DUNDi might be the way to go, or if not, maybe point me in the right direction. snip I wonder if one could use a realtime setup and store the registrations in a common database. I believe I read that is how one shares them between Kamailio and Asterisk - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging into queue homed off remote system
Greetings all, I have an interesting problem I am trying to work around. I currently have 2 * servers running in separate offices, using IAX2 to trunk between them, and queues in our main office. I'll call them Office_A and Office_B. I use Polycom 501s with a primary and secondary server, the primary being whichever server is local to them. What I'm finding is that when a user logs into the queue on Office_A, if their primary server is set to Office_B, the IAX trunk ends up getting logged into the queue, which is no good. I tried using a SIP trunk to make the connection, but I either end up with a digest user mismatch or a Forbidden Auth. After playing around with insecure=port,invite and fromuser=user I was able to log into the queue being homed off Office_B, but the agent logged into the queue was the SIP trunk. Below is the relevant parts of sip.conf, any suggestions are welcome. Office_A sip.conf [general] Register = office_a:sec...@10.10.40.118/office_b [office_b] fromuser=office_a username=office_a type=peer secret=secret context=incoming host=dynamic insecure=port,invite Office_B sip.conf [general] Register = office_b:sec...@10.10.40.118/office_a [office_a] fromuser=office_b username=office_b type=peer secret=secret context=incoming host=dynamic insecure=port,invite Thanks, Bryan STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
